Spaces:
Runtime error
Runtime error
File size: 6,885 Bytes
d3e0df2 3bd5d2b 1f63fcf d3e0df2 ee44eab 1f63fcf ee44eab 1f63fcf ee44eab d3e0df2 ee44eab 87505e7 ee44eab 9758654 ee44eab 3bd5d2b ee44eab 00e31d2 ee44eab 1f63fcf 9758654 24fab16 ee44eab af74e64 ee44eab af74e64 ee44eab e91d345 ee44eab 1f63fcf 3bd5d2b 1f63fcf 1923ff8 9758654 1923ff8 1f63fcf d3e0df2 1923ff8 ee44eab 1923ff8 ee44eab 1923ff8 9758654 ee44eab 1f63fcf 9758654 1923ff8 1f63fcf 9758654 1f63fcf ee44eab 1f63fcf 9758654 1f63fcf 9758654 |
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 |
import base64
import os
from functools import partial
from multiprocessing import Pool
import gradio as gr
import numpy as np
import requests
from processing_whisper import WhisperPrePostProcessor
from transformers.models.whisper.tokenization_whisper import TO_LANGUAGE_CODE
from transformers.pipelines.audio_utils import ffmpeg_read
title = "Whisper JAX: The Fastest Whisper API ⚡️"
description = """Whisper JAX is an optimised implementation of the [Whisper model](https://huggingface.co/openai/whisper-large-v2) by OpenAI. It runs on JAX with a TPU v4-8 in the backend. Compared to PyTorch on an A100 GPU, it is over [**70x faster**](https://github.com/sanchit-gandhi/whisper-jax#benchmarks), making it the fastest Whisper API available.
Note that using microphone or audio file requires the audio input to be transferred from the Gradio demo to the TPU, which for large audio files can be slow. We recommend using YouTube where possible, since this directly downloads the audio file to the TPU, skipping the file transfer step.
"""
API_URL = os.getenv("API_URL")
API_URL_FROM_FEATURES = os.getenv("API_URL_FROM_FEATURES")
article = "Whisper large-v2 model by OpenAI. Backend running JAX on a TPU v4-8 through the generous support of the [TRC](https://sites.research.google/trc/about/) programme. Whisper JAX [code](https://github.com/sanchit-gandhi/whisper-jax) and Gradio demo by 🤗 Hugging Face."
language_names = sorted(TO_LANGUAGE_CODE.keys())
CHUNK_LENGTH_S = 30
BATCH_SIZE = 16
NUM_PROC = 16
FILE_LIMIT_MB = 1000
def query(payload):
response = requests.post(API_URL, json=payload)
return response.json(), response.status_code
def inference(inputs, language=None, task=None, return_timestamps=False):
payload = {"inputs": inputs, "task": task, "return_timestamps": return_timestamps}
# langauge can come as an empty string from the Gradio `None` default, so we handle it separately
if language:
payload["language"] = language
data, status_code = query(payload)
if status_code == 200:
text = data["text"]
else:
text = data["detail"]
if return_timestamps:
timestamps = data["chunks"]
else:
timestamps = None
return text, timestamps
def chunked_query(payload):
response = requests.post(API_URL_FROM_FEATURES, json=payload)
return response.json()
def forward(batch, task=None, return_timestamps=False):
feature_shape = batch["input_features"].shape
batch["input_features"] = base64.b64encode(batch["input_features"].tobytes()).decode()
outputs = chunked_query(
{"batch": batch, "task": task, "return_timestamps": return_timestamps, "feature_shape": feature_shape}
)
outputs["tokens"] = np.asarray(outputs["tokens"])
return outputs
if __name__ == "__main__":
processor = WhisperPrePostProcessor.from_pretrained("openai/whisper-large-v2")
pool = Pool(NUM_PROC)
def transcribe_chunked_audio(inputs, task, return_timestamps):
file_size_mb = os.stat(inputs).st_size / (1024 * 1024)
if file_size_mb > FILE_LIMIT_MB:
return f"ERROR: File size exceeds file size limit. Got file of size {file_size_mb:.2f}MB for a limit of {FILE_LIMIT_MB}MB.", None
with open(inputs, "rb") as f:
inputs = f.read()
inputs = ffmpeg_read(inputs, processor.feature_extractor.sampling_rate)
inputs = {"array": inputs, "sampling_rate": processor.feature_extractor.sampling_rate}
dataloader = processor.preprocess_batch(inputs, chunk_length_s=CHUNK_LENGTH_S, batch_size=BATCH_SIZE)
try:
model_outputs = pool.map(partial(forward, task=task, return_timestamps=return_timestamps), dataloader)
except ValueError as err:
# pre-processor does all the necessary compatibility checks for our audio inputs
return err, None
post_processed = processor.postprocess(model_outputs, return_timestamps=return_timestamps)
timestamps = post_processed.get("chunks")
return post_processed["text"], timestamps
def _return_yt_html_embed(yt_url):
video_id = yt_url.split("?v=")[-1]
HTML_str = (
f'<center> <iframe width="500" height="320" src="https://www.youtube.com/embed/{video_id}"> </iframe>'
" </center>"
)
return HTML_str
def transcribe_youtube(yt_url, task, return_timestamps):
html_embed_str = _return_yt_html_embed(yt_url)
text, timestamps = inference(inputs=yt_url, task=task, return_timestamps=return_timestamps)
return html_embed_str, text, timestamps
microphone_chunked = gr.Interface(
fn=transcribe_chunked_audio,
inputs=[
gr.inputs.Audio(source="microphone", optional=True, type="filepath"),
gr.inputs.Radio(["transcribe", "translate"], label="Task", default="transcribe"),
gr.inputs.Checkbox(default=False, label="Return timestamps"),
],
outputs=[
gr.outputs.Textbox(label="Transcription"),
gr.outputs.Textbox(label="Timestamps"),
],
allow_flagging="never",
title=title,
description=description,
article=article,
)
audio_chunked = gr.Interface(
fn=transcribe_chunked_audio,
inputs=[
gr.inputs.Audio(source="upload", optional=True, label="Audio file", type="filepath"),
gr.inputs.Radio(["transcribe", "translate"], label="Task", default="transcribe"),
gr.inputs.Checkbox(default=False, label="Return timestamps"),
],
outputs=[
gr.outputs.Textbox(label="Transcription"),
gr.outputs.Textbox(label="Timestamps"),
],
allow_flagging="never",
title=title,
description=description,
article=article,
)
youtube = gr.Interface(
fn=transcribe_youtube,
inputs=[
gr.inputs.Textbox(lines=1, placeholder="Paste the URL to a YouTube video here", label="YouTube URL"),
gr.inputs.Radio(["transcribe", "translate"], label="Task", default="transcribe"),
gr.inputs.Checkbox(default=False, label="Return timestamps"),
],
outputs=[
gr.outputs.HTML(label="Video"),
gr.outputs.Textbox(label="Transcription"),
gr.outputs.Textbox(label="Timestamps"),
],
allow_flagging="never",
title=title,
examples=[["https://www.youtube.com/watch?v=m8u-18Q0s7I", "transcribe", False]],
cache_examples=False,
description=description,
article=article,
)
demo = gr.Blocks()
with demo:
gr.TabbedInterface([microphone_chunked, audio_chunked, youtube], ["Transcribe Microphone", "Transcribe Audio File", "Transcribe YouTube"])
demo.queue(max_size=3)
demo.launch(show_api=False)
|